Full duplex wireless communication systems are well known and include various types of systems, such as cellular telephone systems, personal communication systems, integrated systems, such as the “MOTOROLA” “iDEN” system, and real-time video systems. Such wireless systems are known to include a system infrastructure and communication devices constructed and programmed to operate in the respective system. The system infrastructure includes fixed network equipment, such as base transceiver sites (BTSs), system controllers (e.g., base site controllers (BSCs)), switching centers, routers, communication links, antenna towers, and various other known infrastructure components.
Certain wireless communication systems, such as digital cellular systems and the “MOTOROLA” “iDEN” system utilize voice compression techniques, such as vector sum-excited linear predictive (VSELP) encoding, advanced multi-band excitation (AMBE) encoding or code-excited linear predictive (CELP) encoding, to more efficiently utilize available frequency spectrum. In such systems, each BSC and each wireless communication device include one or more transcoders that encode (compress) pulse code modulated (PCM) or other digitally sampled voice signals received from the audio portion of the infrastructure (e.g., from a mobile switching center (MSC)) or communication device, respectively, and decode (decompress) received compressed voice signals received from the radio frequency portion of the infrastructure (e.g., BTS) or communication device, respectively. Each transcoder includes one or more digital signal processors (DSPs) and associated control circuitry and operating software. With current DSP technology, each DSP can support one or more two-way, full duplex audio communications.
To compress incoming PCM audio signals, the infrastructure transcoder typically encodes a predetermined amount of audio and packetizes the encoded audio to comply with the transmission format of the outgoing communication link. The amount of audio encoded by the transcoder corresponds to the radio transmission access protocol employed by the BTS. For example, the BSC transcoder of the “iDEN” system VSELP encodes ninety (90) milliseconds (ms) of audio in two forty-five (45) millisecond transmission frames or three thirty (30) millisecond transmission frames (depending on whether the BTS transmission rate is full rate or half rate) and packetizes the transmission frames into a single information packet in accordance with a high data level link control (HDLC) protocol utilized on the communication link between the BSC and its associated BTSs. The “iDEN” BTS employs a time division multiple access (TDMA) transmission protocol in which each time slot supports ninety (90) milliseconds of information (compressed audio or data). Therefore, the ninety milliseconds of compressed audio in each HDLC information packet coincides with the amount of compressed audio that may be transmitted from the BTS in a single time slot.
The timing at which the BSC communicates compressed transmission frames to a BTS also depends on the particular radio transmission access protocol employed by the BTS. Such timing is particularly important when the access protocol is time-based, such as the TDMA protocol utilized in the “iDEN” system and other digital cellular systems (e.g., the Global System for Mobile Communications (GSM)). For proper transmission from a BTS that utilizes a TDMA protocol, the BTS and BSC must be synchronized such that the BSC conveys an information packet, which may include multiple transmission frames, to the BTS prior to the beginning of the transmission time slot allocated for transmitting the packet. Optimally, the packet should arrive at the BTS just early enough to enable the BTS to process the packet (e.g., modulate, filter, upconvert, and amplify) before the beginning of the transmission time slot to minimize the amount of buffering or storage of compressed audio that must occur at the BTS and, therefore, minimize the gaps or choppiness in audio perceived by the user of the recipient communication device.
However, due to various delay mechanisms within the infrastructure, and even within the transcoders themselves, a fixed synchronization between BSC and BTS is not possible. Therefore, BSCs and BTSs typically employ a synchronization protocol in an attempt to maintain synchronization in view of the various delays. Under such protocols, the BTS instructs the BSC to adjust its packet transmission time based on the historical arrival of information packets from the BSC for a particular communication.
BSCs typically service multiple BTSs in full duplex wireless systems. Consequently, each BSC commonly includes a router that directs encoded audio from the transcoder to the appropriate BTS. When a deterministic transport protocol, such as a circuit-switched protocol, is used between the router and the BTS, the router introduces a determinate, but not necessarily fixed, delay which can be taken into account by the BTS when instructing the BSC to transmit a particular information packet in accordance with the synchronization protocol. However, packet switching is becoming more popular these days to increase the effective bandwidth of the communication path between the BSC and the BTS.
When a packet-switched transport protocol is used to communicate information from the BSC to the BTS, the delays introduced by the router become non-deterministic and may produce wide variances in arrival times of information packets at the BTS. In an attempt to account for such wide variances, prior art BTSs estimate the router delay and request transmission of packets from the transcoder based on the estimate in accordance with their respective synchronization protocols. However, notwithstanding such estimation by the BTS, the nondeterministic nature of the BSC-to-BTS transport can result in receipt of a packet by the BTS after the BTS transmission time for the packet has expired. That is, the indeterminate delays of the router in a nondetermistic system can result in missed packet transmissions and poor signal quality (e.g., choppy audio) as perceived by a user of a wireless communication device.
In addition to introducing variable delays, a non-deterministic packetized transport employed between the BSC and the BTS provides no order with which the router transmits packets received from the transcoder. Such a lack of order can introduce undesirable delays, and therefore poor perceived signal quality, in communications in which synchronization between the BSC and BTS is already established. For example, if the router receives a packet for a first full-duplex communication that is just commencing and in which synchronization is not yet established between the BSC and the BTS, and another packet for a second full-duplex communication that is ongoing and in which synchronization has been established between the BSC and the BTS, the router may communicate the first communication's packet to the BTS before communicating the second communication's packet, thereby introducing an undesired delay in the conveyance of the second communication's packet and degrading perceived quality of the second communication.
Therefore, a need exists for a method and apparatus for improving signal quality of transmitted information as perceived by a user of a wireless communication device that accounts for variable delays introduced by a BSC's router when a non-deterministic packetized transport is used to communicate information between the router and the BTS, and that appropriately prioritizes transmissions from the BSC to its respective BTSs to minimize delays and thereby substantially improve the quality of communications as perceived by users of recipient wireless communication devices.